Method for quantitatively measuring a hearing defect

ABSTRACT

A method is described for measuring the hearing deficit of a person preparatory to providing a hearing aid designed in accordance with analysis of the deficit, wherein frequency bands over which hearing level is unacceptable are determined, the bands are then shifted in harmonic sustaining manner to bands over which hearing level is acceptable, and thereafter over all frequency bands parameters necessary to shape, amplify, compress and adaptively filter background noise in continuously variable and adaptive fashion so as to produce an aurally optimized signal are adjusted and recorded.

This is a divisional of application Ser. No. 371,584, filed Apr. 26,1982, now U.S. Pat. No. 4,419,544, which is a divisional application ofSer. No. 144,395, filed Apr. 28, 1980, now U.S. Pat. No. 4,366,349.

BACKGROUND

This invention relates generally to hearing aids and more particularlyconcerns a method and apparatus for improving hearing communication inpersons with various hearing deficits. The invention will be disclosedin connection with a generalized signal processing unit which restoresmeaningful neural patterns to the auditory recognition centers of thebrain in a manner which enables persons and other higher animals torecognize a variety of sounds including human speech.

The present invention is derived from an understanding of thephysiological processses of the normal hearing apparatus, and the mannerin which the auditory system is compromised in conditions of hearingloss. The hearing apparatus in humans and higher animals consists ofmechanical parts which transmit acoustic pressure waves from the air tothe sensory neural network. The function and pathologies of theconductive portion of the hearing apparatus are well known up to thepoint of the mechanical-to-neural transduction site.

The mechanical-to-neural transformation occurs within the cochlea alonga structure called the basilar membrane. This membrane demonstratesexcursions which have a position-to-tone relationship. A vast network ofneural cells which are sensitive to displacement are disposed in rowsalong the length of the membrane. The membrane thus performs a spectralanalysis on the incoming sound, directing various portions of thefrequency spectrum along specific neural channels. The specific neuralcells which are involved in this transduction process are called theinner and outer hair cells. The inner hair cells are in primaryrelationship with the nerves which ascend toward the brain while theouter hair cells are in primary relationship with nerves which carryinformation down from the brain. The exact functional role of these twoneural hair cells types has not been conclusively proven. Applicantpostulates that the role of the inner hair cell is that of a purereceptor, and that this cell initiates electrical signals along itsneural channel to the brain when it is displaced as a result of themotion of the basilar membrane. It is further postulated that the outerhair cell exerts a control function on the inner hair cell, specificallythat it alters the threshold of displacement which must be exceededbefore an electrical response can be initiated along a given neuralpathway. Hence the inner and outer hair cells comprise the second andmodifying functions of an adaptive control system. It follows that thereis an adaptive portion to the hearing system, that it is responsible forthe high degree of pitch discrimination in humans and higher animals,that disease processes affect both the receptor inner hair cells and theadaptive controller outer hair cells, and, finally, that the structureplaces physiological constraints on the remedial signal processing whichcan be performed by any hearing aid.

Conventional hearing aids serve as amplifiers of the auditory signal.The presentation of an amplified signal does provide a means forrestoring the conductive functions of the ear. However, amplificationalone is of only marginal help if the mechanical-to-neural structure hasbeen damaged. In these cases the neural channels that carry specificfrequencies to the brain and their contribution to the adaptivityfunction have been lost. Under extreme amplification, the excursion ofthe basilar membrane will be broad enough to excite adjacent hair cells,and if these cells exist, a signal will be sent to the brain. However,the amplification does nothing to restore adaptivity, and the functionof identifying important sounds from background noise can occur onlywith conscious effort at higher levels of the brain. Applicantpostulates that this effort is fatiguing and is the reason why theadjustment period for a hearing aid user is difficult. In more advancedisease of the sensor-neural network, adjacent channels may not bepresent for stimulation by signals which are amplified to levels whichcan be physically damaging to the auditory structures. In these casesthe portion of the frequency spectrum which has no neural channels mustbe readdressed to other still viable neural channels, and the adaptivityfunction belonging to the lost frequency portion must be performedexternal to the ear.

A variety of hearing aid devices have been proposed for addressing theproblem of sensory-neural deafness. The methods and means of theprevious methods have utilized band shifting of frequencies, varioustransformations of the auditory signal, alteration of the format loci,and amplitude compression schemes. The methods fail to recognize theinseparability of the adaptive and receptor functions, and only addressa part of the necessary signal processing functions which must beperformed. furthermore, the previous methods fail to recognize thehighly individual nature of the hearing loss in any patient, and theneed for the minimal but generalized signal processing to restorerecognizable auditory patterns in the auditory cortex of the brain.

Accordingly, it is an objective of the present invention to provide newand useful methods and means for restoring recognizable auditorycommunication to persons having various deficits in their auditorysensor apparatus.

It is another objective of the present invention to define methods andmeans for the characterization of residual hearing function in terms ofthe spectral and adaptive portions of a signal processing system.

SUMMARY OF THE INVENTION

In accordance with one aspect of the invention, a signal processingapparatus is provided which receives a first acoustic signal andgenerates a second acoustic signal with a modified pattern in responseto the first signal. The apparatus includes means for initiallyreceiving the first acoustic signal and generating a representativeprocessed signal. Portions of this generated signal which do not exceeda predetermined pattern recognition function are then suppressed by anadaptive noise cancelling means. The signal produced by the adaptivenoise cancelling means is then amplified and converted into a modifiedacoustic signal.

According to a further aspect of the invention, a hearing aid includes ameans for generating a processed signal representative of a receivedacoustic signal. This generated signal is applied to a frequency bandseparating means which divides the signal into at least two separatedsignals according to frequency. At least one of these separated signalsis directed to a means for transposing the frequency content to higheror lower frequency ranges while maintaining the harmonic relationshipcharacterizing the band separated signal. Further means are provided fortemporally matching the band separated signals and for summing thetemporally matched signals into a resultant signal which is, in turn,converted into a modified acoustic signal.

Yet another aspect of the invention relates to an apparatus for testingheating capabilities in humans. The apparatus includes a means forreceiving an acoustic signal and generating a responsive andrepresentative processed signal. An adaptive noise cancelling meanssuppresses portions of the generated signal which do not exceed apredetermined pattern recognition function. This noise cancelling meansalso includes means to variably adjust the characteristics of thepattern recognition function as well as means to restrict the frequencycontent upon which the pattern recognition is performed. A variablefrequency band separating means divides the correlated signal into aplurality of band limited output signals, each of which is directed to acircuit path with a non-linear amplifier for automatically greateramplifying the lower intensity portions of the signal while lesseramplifying its higher intensity portions. The non-linear characteristicof the amplifier is variable. The testing apparatus further includesmeans for selectively and variably transposing the frequency of theseparated signals while maintaining the harmonic relationships of saidseparated signals.

BRIEF DESCRIPTION OF THE DRAWINGS

The above and other objects, features, and advantages of the presentinvention will become apparent from a consideration of the followingdetailed description presented in connection with the accompanyingdrawings in which:

FIG. 1 shows a schematic depiction of the normal signal processingfunctions which comprise the human auditory system.

FIG. 2a depicts the frequency spectrum of a typical auditory signal anda schematized representation of the effect of a neural hearing loss onthe auditory spectrum.

FIG. 2b depicts the movement of portions of the auditory spectrum fromregions of hearing loss to higher and lower adjacent frequencies.

FIG. 2c depicts the summation of the so moved portions of signal withfrequency portions of the signal which is addressed to viable portionsof the neural auditory system.

FIG. 3 is a schematic of a generalized signal processing apparatus madein accordance with the principles of the present invention.

FIG. 4 is a detailed schematic of one embodiment of a practicalelectronic circuit for the nonlinear amplification element of theapparatus.

FIG. 5 is a detailed schematic of one embodiment of a practicalelectronic circuit for realizing the harmonic spectral shifting elementof the apparatus.

FIG. 6a depicts a typical auditory signal of repetitive time period andvarying amplitudes in the presence of a competing but lower amplitudeperiodic noise signal.

FIG. 6b depicts the signal presented in FIG. 6a as delayed in time byone major time period of the auditory signal.

FIG. 6c depicts the noise cancelling effect of summing a signal delayedby one major time period of the auditory signal with a non-delayedsignal of similar shape.

FIG. 7 is a detailed schematic of one embodiment of a practicalelectronic circuit for realizing the noise cancelling adaptive filterelement of the apparatus.

FIG. 8 is a detailed schematic of one embodiment of a practicalelectronic circuit for realizing the segment deglitch window element ofthe apparatus.

DESCRIPTION OF THE PREFERRED EMBODIMENT

Referring now to the drawings and to FIG. 1 in particular, a schematicdepiction of the normal signal processing functions of the humanauditory system is shown. An auditory signal 10 is applied to theexternal ear and the conductive network of the middle ear as representedby box 12. In this portion of the system the acoustic pressure waveproduced by the auditory signal is converted into movement of themechanical structures of the middle ear. This mechanical movement isthen transmitted to the basilar membrane of the inner ear depicted bybox 14 in the illustration of FIG. 1. This mechanically responsivemembrane 14 receives mechanical stimulus from the middle ear structuresand causes a spectrally selective excitation of neural cells 18 commonlyknown as inner hair cells which are disposed along its expanse. Onexcitation, a portion of these cells 18 transmit electrical impulsesalong ascending neural pathways 20 to the brain 22. A second portion ofthe neural signal 23 so generated, and a delayed, evoked neural signal27 descending from the brain the brain produce an effect on yet anotherset of neural cells 24 commonly known as outer hair cells which are alsolocated on the basilar membrane 14. The effect on the outer hair cells24 produced by the combined neural and mechanical signals causes themodification indicated by path 25 of the local electro-chemicalenvironment of inner hair cell 18. The electro-chemical environmentaffects the threshold at which the inner hair cell responds tomechanical stimulus.

Mechanical motion of the basilar membrane in response to motion of themiddle ear structures initially stimulates a plurality of both inner andouter cell groups located at positions of maximal excursion along themembrane. The portion of membrane which is elevated sufficiently tocause initial firing of the inner hair cells is relatively broad, and byitself cannot account for the tonal acuity which the auditory systemdemonstrates. A modification of the electro-chemical environment, asdepicted as the process arrow 25 in the illustration, alters thethreshold of the inner hair cells 18 and results in the local sharpeningof the neural spectrum. This sharpened neural response establishes thebandwidth of pitch perception. A neural input 26 to the outer hair cells24 acts in concordance with mechanical input to determine thestimulation of those cells. Signals on the neural input 26 are thesummation of neural signals arising from the local inner hair cells 18,from other groups of hair cells 28, and from the brain 22. The othergroups of hair cells 28 are located at points along the basilar membrane14 which correspond to frequency spectral differences of approximatelyone-sixth of an octave. Applicant postulates that the contribution ofthe other groups of cells is to modify the activity of the outer haircells 24, and results in enhanced sharpness.

The mechanical-to-neural portion of ear perceives bands of excitationwhich correspond to the hair cell groups in a manner analogous to thefunction of a spectrum analyzer. Thus the ear is comprised of aplurality of contiguous receptor/modifier channels each having primaryresponsibility for the interpretation of a particular segment or band ofthe frequency spectrum. Because the mechanical-to-neural transductionprocess is frequency responsive, a loss of selected hair cell grops willresult in a correspondingly selective loss in the audio frequency range.

FIG. 2a depicts the frequency spectrum of a typical auditory signal asthe solid line 31 with its intensity plotted on the ordinate versus itsfrequency plotted on the abscissa. The dashed line 32 of FIG. 2arepresents the filter characteristic of a pathological ear with itslow-valued portion 33 corresponding to the loss of hair cell groups. Inthe presence of pathologies affecting the hair cell groups, thefunctions of these components are lost. Hence both the receptor functionof the inner hair cells and the modifying function of the outer haircells fail. The result is twofold: first the primary signal cannot bepresented to a viable neural ascending channel, and second thesharpening of the signal cannot occur. Under such pathologicalconditions, the lost original portion 36 of the frequency spectrumcannot be recovered by simple amplification because no viable neuralchannels exist for its reception.

FIG. 2b demonstrates the movement of the lost information 36 to viableneural channels. The information in the lost band is shifted upwardand/or downward in frequency to be readdressed to the brain via haircell groups which have remained intact. The so-called lost information36 which has been so shifted in a physiologically appropriate manner isthen added to the viable portions of the receptor spectrum 34 as shownin FIG. 2c.

A schematic depition of a generalized signal processing bearing aidapparatus is illustrated in FIG. 3. An audio signal is received by amicrophone 40 which in turn provides an electrical signal to bufferamplifier 42. The output signal from buffer amplifier 42 is directedinto an adaptive noise cancelling element 44 for the purpose of deletingor suppressing portions of the auditory signal which do not correlate orhave a coincident pattern relationship with the strongest dominantfrequency residing in a predetermined range. The resulting output signalproduced by the adaptive noise canceller 44 is then applied to a fastattack and slow release type automatic gain control amplifier 45, theoutput of which is directed to a frequency band separating network 46.This network divides the signal by means of variable bandpass filters 48according to frequency content and produces a plurality of signals eachcontaining a variably selected portion of the frequency spectrum. Eachof these variably selective portions of the signal spectrum is theninput to one of a plurality of signal modification networks 50. Only oneof the signal modification networks is shown in detail in the drawings,although the other represented networks 50 contain detail elementsidentical to those described in the sequel. The signal modificationnetwork 50 contains a nonlinear amplifier element 52, a harmonicfrequency transposer element 54, a bond filter element 55, and againshaping amplifier element 56. The output of each of the signalmodification networks 50 is input to a summing amplifier 59 throughvariable attenuating elements associated with each signal modificationnetwork 50 and collectively identified on the illustration by thenumeral 58. Output from the summing amplifier 59 is directed to a poweramplifier 70 having suitable impedance characteristics to match itsoutput to an earpiece/speaker 72 for transmission to the patient'sdefective auditory system.

FIG. 4 illustrates one example of an amplifier system which can serve asthe nonlinear amplifier element 52 identified in FIG. 3. The illustratednonlinear amplifier element 52 is a double sided logarithmic type andincludes an input decoupling capacitor 73 for removing the d.c. offsetfrom the signal. The resulting signal is then divided into positive andnegative portions by diodes 74 and 77 respectively. The positive portionof the signal is then caused to be processed through an amplifier stage76 employing an NPN transistor feedback element 75 operating in itsnonlinear region. Appropriate passive elements useful to the operationof the amplifier stages are also indicated in the illustration. Alogarithmic output from the amplifier stage 76 is thus obtained for thepositive portion of the signal. The negative portion of the signal issimilarly processed by the logarithmic amplifier stage 79 employing aPNP transistor feedback element 78 operating in its nonlinear region.The positive and negative logarithmic output signals are then caused tobe added together in the summing amplifier 80 to provide input for theharmonic transposition element shown in detail in FIG. 5.

FIG. 5 illustrates one embodiment of a harmonic transposition element 54for the selective multiplication or division of all frequencies by aconstant value. The movement of all frequecies by a multiplicativeprocess retains the harmonic relationships which are present in theoriginal signal. Because of the importance of the relationship ofharmonics in the recognition of speech and other significant acousticpatterns, and because of the postulated role of harmonic contribution inthe mechanical to neural transduction process at the hair cell level,only harmonic transposition is suitable for retention of informationcontent in a form recognizable to the brain.

In the illustrated form depicted in FIG. 5, the output of the nonlinearamplifier 52 is applied to a time axis dividing, sampled signal,frequency harmonic transposition system 54 through the agency of aninput buffer and offset preamplifier 82. This element isolates thesignal from previous stages and also adds a d.c. voltage. The d.c.offset voltage is necessary for the processing of the signal through amulti-element, storage component such as a conventional dual serialanalog delay element 83. One such element is an SAD 1024 manufactured bythe Reticon Corporation of Sunnyvale, Calif. The isolated and offsetsignal is then caused to be input to the discrete time serial analogmemories contained in the serial analog element 83. The movement of thesignal through each serial memory of the analog delay element 83 occursin a "bucket brigade" fashion, with the signal being passed from onememory cell or "bucket" to the next on the command of a clock pulse.Independent clock pulses are provided to each memory to achievedifferent delays, or movements of "information buckets", in each memorydelay line.

In operation, memory `A` (84) is driven by timing pulses from a firstclock `F` 87 arriving through the agency of clock switch 92. Thebuffered, offset signal is input to memory `A` as a string of sampleddata as long as the fast clock interruptor switch 95 remains closed. Thedelayed output of memory `A` is directed to ground through the agency ofaudio signal switch 90. The memory `A` (84) as depicted in FIG. 5 issaid to be in its input cycle. The contents of memory `A` (84) thusrepresents a sampling of the signal with samples being taken at thefrequency of clock `F` 87. During the immediately previous positioningof the audio switch 90 and clock switch 92, memory `B` (85) was filledin an identical manner. In the depicted position of the clock switch 92and the audio switch 90, memory `B` (85) is driven by timing pulsesarriving from a second clock `S` 86. The contents of memory `B` (85) areoutput to the low pass filter 96 through the agency of audio switch 90.The memory `B` (85) as depicted in FIG. 5 is said to be in its outputcycle. After clock `S` has generated the number of pulses exactly equalto the number of cells in the memory, primary switch driver 89 causesaudio switch 90 and clock switch 92 to be moved into their other contactpositions. This reverses the functions of the memories. In this way amemory may be loaded at the fast clock rate from clock `F` 87 andunloaded at the relatively slower clock rate from clock `S` 86, theresult being an apparent division of all frequencies of the signal by aconstant amount corresponding to the ratio of the slow to fast clockrates. The unload clock `S` in each of the signal modification networks50 are preferably synchronized and output at the same rate to providetemporal matching among the signals output from the various signalmodification networks 50.

In general a portion of the signal must be discarded during thefrequency division process because no alteration in the temporal periodoccurs; i.e. the inverse relationship between time and frequency of thesignal is violated. As a result, discontinuities normally arise betweenthe signal segments at each switching of the memory function. Circuitrycontained in the segment deglitch window element 94 senses theattributes in the signal portions prior to switching, and provides meansfor minimizing such discontinuities. The segment deglitch window element94 causes a moveable "window" of sample time by variably opening clockinterruptor switch 95. The moveable time "window" is caused to stop whenconditions are met which minimize discontinuities arising from theswitching of the memory functions.

In a similar manner, well-known memory recirculation schemes may beemployed to provide for a continuous signal generation and a variablymoving window when a multiplication of the signal frequency is to beaccomplished, i.e. when the ratio of the clock rates is greater thanunity.

There are upper and lower restrictions on the delays which may beimposed on the audio circuit for hearing aid applications. Theserestrictions dictate the ranges for the clock frequencies and,secondarily, they restrict the range for the number of cells containedin the memories. These restrictions arise from sampling theory,linguistic, auditory sensitivity, and temporal bases. Signal theoryrequires that the sampling rate be at least twice the frequency of thehighest frequency contained in the auditory signal and also that atleast one complete cycle of the lowest frequency of the auditory signalbe retained in the memory. Linguistic sensibility demands that theamount of information which is discarded be somewhat less than a phonemetime period. Auditory sensitivity requires that the sampling frequencylie above the frequency range for any individual, and it also requiresthat the discontinuities occur at the broadest time intervals. Finally,phonemic temporal restrictions require that the transposition beaccomplished in as nearly "real time" as is possible. This lastrestriction is imposed to preserve lip synchronization, and isparticularly important to persons with hearing deficit. A suitable setof parameters for the depicted circuit provides an intermediate delay of10 to 60 milliseconds using clock frequencies in the 15 kHz range and512 memory cells. These parameters insure that within a linguisticallydistinct period there will be temporal matching of the signal.

FIGS. 6a, 6b, and 6c are an expository representation of the function ofthe suppression of noise from "meaningful" auditory signals. Thepreselection of a "meaningful" auditory signal is postulated on themotion that speech arises from a vibrating source having a fundamentalfrequency of less than 500 Hz, and that the modifications or formantswhich convert the vibrations into linguistic patterns are correlatedwith the fundamental or a low harmonic of the vibrating sound source. Ingeneral, the competing noise will not have periods which coincide withthe period or multiples of the period of the "meaningful" auditorysignal.

In the representation of FIG. 6a, a "meaningful" auditory signal, asshown by the heavy line 98a, consists of four periods `a`, `b`, `c`, and`d` of primary information plus some uncorrelated noise signalrepresented by the light, sinusoidal line 98b. A time-shifted version ofthe signals 98a and 98b is depicted in FIG. 6b wherein the time shiftcorresponds to exactly one time period of the auditory signal. Thedepicted signal retains three of the original four periods of theauditory signal. If the signals of FIG. 6a and FIG. 6b are addedtogether they form a new signal as shown in FIG. 6c which reinforces theprimary signal and all signals which have time correlates with it, whilesimultaneously suppressing all noncorrelated noise signals.

FIG. 7 illustrates one embodiment of an adaptive noise canceller element44 for the suppression of noise signals from meaningful auditory input.The illustrated means is based on the correlative principlesdemonstrated in FIGS. 6a, 6b, l and 6c. Within the element arefunctional blocks which constitute an autocorrelation group 102, acorrelation peak group 112, and a correlative canceller group 126. Inthe depicted preferred embodiment, autocorrelation group 102, and peakcorrelation group 112 together constitute a pattern recognition means.Said means enables the periodic "locking" of a quasi-static signal witha delayed version of itself in the correlative canceler group 126. Ifthe signal is a pure sinusoid, the elements 102, 112 and 126 perform thefunction of a simple phase locked loop. Similarly, if the patternrecognition requirements of the system are relaxed to require therecognition of only a single frequency, the depicted circuitry may bedegenerated to constitute a simple phase locked loop.

Operatively, the auditory signal which has been processed by bufferamplifier 42 of FIG. 3 is caused to enter a splitting junction 104 whereit is directed to both the autocorrelative group 102 and the correlativecanceller group 126. Within the autocorrelation group 102 the auditorysignal is applied to a bandpass buffer amplifier/filter 105 andthereafter to both a delay line 108 and to the `X` input of a multipliermodule 110. The buffer amplifier/bandpass filter facilitates theselection of the correlation on a particular frequency portion of theoriginal signal. The delay line 108 consists of a number of sequentialstorage elements which cause the signal to be passed from element toelement on a clock signal. The clock signal is provided by a voltagecontrolled oscillator 106 whose output frequency is caused to varylinearly in response to the output of a sawtooth waveform generator 107.The portion of the signal passing through the delay line 108 is directedto the `Y` input of the multiplier module 110. The product of thenondelayed portion of the signal and the portion of the signal which hasundergone the variable delay is then directed to an integrator circuit118. Output from the integrator 118 corresponds to a short-timeautocorrelation function: ##EQU1##

The time period of the correlation function is determined by thesawtooth characteristics of the voltage which controls the oscillator106 and the integrator circuit switch 119 characteristics. In thepreferred embodiment, the period is set to substantially correspond tothe fundamental and first harmonic frequencies of human speech. Thecorrelation function will achieve a positive maximum value when thedelayed signal has been retarded by one period as depicted in FIG. 6b.

The correlation signal is then directed to the correlation peak group112 where the time period between repetitive portions of the auditorysignal is determined. The correlation signal is directed through diode117 to a positive peak detector module 120. The peak detector module 120determines the value of the highest power over a composite time period.This composite time period is determined by the correlation functionperiod, the integrator time constant, and a third time restrictionprovided by a refresh clock 121. The refresh clock 121 prohibits thepeak energy which occurs during one portion of a word from beingcompared with a lesser power exhibited at a linguistically temporallydistant time. For example, the time between the sounds "pow" and "er" inthe word "power" would be functionally separated by the refresh clock121. Output from the positive peak detector 120 and output from theintegrator 118 are directed into the input of a comparator circuit 122which produces a logical HI state when the values of the peak detector120 and the integrator 118 are nearly identical. A hysteresis type ofcomparator suppresses the tendency to change the states of the output inan unstable fashion. The comparator 122 produces a logical valued outputthrough the agency of a NAND device. In operation, when the currentvalue of the output of the autocorrelation group 102 and the value ofthe peak detector 120 are nearly equal, the autocorrelation function isnear its optimal time delay. The resulting logical HI state thusproduced on the comparator 122 is used to "hold" the instantaneous valueof the sawtooth waveform generator 107 in the sample and hold circuit127. This voltage then holds the delay line clock 128 at a constantfrequency and, in turn, produces a delay in the portion of the auditorysignal passing through delay line 130 equivalent to one period of theprimary frequency of the auditory signal. The upper and lower clockfrequencies are predermined by the sound environment and individualhearing loss characteristics of the hearing aid wearer.

The auditory signal which has been directed from buffer amplifierelement 42 of FIG. 3 into the correlative canceller group 126 firstpasses through buffer amplifier 131 and then is caused to follow twopathways through the circuitry of the correlative canceller group 126.One of these paths goes through the delay line 130 which retards itsprogress by one period of the primary frequency of the auditory signalthrough the means described previously. The other path is a simpleundelayed transmission line. The undelayed and the delayed portions ofthe signal are then caused to pass through balancing resistors 132 afterwhich they are added together in the summing amplifier 134 wherein thesignal attains the characteristics depicted in FIG. 6c.

FIG. 8 illustrates one embodiment of a segment deglitch window element94 for the minimization of transitional discontinuities arising from thejoining of originally discontiguous signal segments. The illustratedmethod is based on the concept of the movement of a time "window" overthe portion of the signal which is to become the next segment loadedinto memory. Within the element are functional blocks which constitutean auxiliary window memory group 140 and a segment deglitch logic group160.

Operatively, the window memory group 140 provides means for determiningthe value of trailing portions of a first signal segment and the valuesof the leading portions of a next signal segment. A sequential memoryelement 144 receives a continuous signal from the buffer offsetamplifier 82 of FIG. 5. The signal is then sampled and caused to advancesequentially through the memory by the agency of clocked pulses arisingfrom clock `F` 87 of FIG. 5. Intermediate "taps" are provided along thememory at the locations of the lowest and second lowest, as well as thehighest and second highest memory addresses. The value of the sampleresiding in each of these addresses is thus available through the agencyof the appropriate tap.

The samples residing in the lowest and second lowest memory addressesare directed to switch 146 where they are each directed to an element ofthe multiple sample and hold network 150. Each of the storage elementsin the sample and hold network 150 will hold the instantaneous value ofits input signal upon the application of the sequence of a logical lowfollowed by a logical high value on the set/reset. The sample and holdelement will continue to hold the said value until a low-high sequenceis again applied. The appropriate logic values are applied alternatelyto pairs of the sample and hold elements through the agency of switch147 which derives its logic values from output of the circuitry ofsegment deglitch logic group 160. The values residing on the storageelements are brought out in alternating pairs through the agency ofswitch 147. The switches 146, 147 and 148 are caused to change state inaccordance with a signal provided by the primary switch driver 89 ofFIG. 5. The output of switch 148 thus provides the last and the next tolast value of the previous signal segment on lines 162 and 164respectively. The values of the first and second samples of the newsignal segment are taken directly from the highest and next highestmemory addresses of auxiliary memory 144. The first value is presentedon signal line 166, and the second value is presented on signal line168.

The segment deglitch logic group 160 utilizes the information providedby the window memory group 140 to perform a hierarchy of logicaldecisions. These decisions determine the starting position of signalsegment which is to follow the signal segment which is immediately inits output function. The decisions provide, in order, that (1) acompletely full complement of samples will be available in the memorywhen it is switched to its output function; (2) the amplitude of thefirst sample of the memory being switched into the output function willsubstantially match the amplitude of the last sample of the memory whichis immediately in its output function; and (3) the slope of the firstsamples of the memory being switched into the output function willsubstantially match the slope of the last samples of the memory which isimmediately in its output function. Additionally, limits are placed onthe amount of time that the conditions (2) and (3) are permitted to bein force.

The values of the last two signal samples of the memory which isimmediately units output function are communicated to the positive andnegative inputs of a subtraction amplifier element 170 for the purposeof calculating the slope of the trailing portion of the identifiedsignal segment. Similarly, the values of the first two signal samples ofthe memory which is preparing for its output function are communicatedto the positive and negative inputs of a subtraction amplifier element172 for the purpose of calculating the slope of the leading portion ofthat identified signal segment. The output from said subtractiveamplifier 172 is compared with the output of said subtractive amplifier170 at the hysteresis comparator 176. The comparator output will be in aLO state only when the slopes are nearly equivalent. The output ofcomparator 176 is inverted in the NAND device 178, the output of whichis directed to the OR device 180. The OR device 180 has another inputarising from ripple counter 184. This lead will cause the OR device 180to produce a logical HI state after a predetermined time, whether or notthe slope matching conditions have been satisfied. The ripple counter184 is driven by pulses arising from clock `F` 87 of FIG. 5, and isreset to a zero count state at each switching of the memory functions byprimary switch driver 89 of FIG. 5.

The amplitude values are compared in a similar manner. The value of theleading sample of the memory being prepared for input is communicated bysignal line 166 to one input of comparator 186. The amplitude of thetrailing sample of the memory in immediately in its output function iscommunicated by signal line 162 to the reference input of comparator186. The output of comparator 186 will be in a LO state when theamplitudes are nearly equivalent. The output of comparator 186 is theninverted in the NAND device 188, the output of which is directed to theOR device 190. The OR device 190 has another input arising from ripplecounter 184. This other input will cause the OR device to produce alogical HI state after a predetermined time, whether or not theamplitude matching conditions have been satisfied.

The outputs from the OR devices 180 and 190 are directed to an ANDdevice 192. A HI output is produced on AND device 192 when the slope andamplitude conditions, or their time default conditions have beenachieved. Output from AND device 192 serves as one of the inputs to afinal AND device 198. The other input to AND device arises fromflip-flop device 194. The flip-flop device 194 will be caused to obtaina HI state upon receiving a pulse from the ripple counter 184. Saidenabling pulse will occur in accordance with a predetermined number ofpulses of clock `F` 87 of FIG. 5 which correspond to the number ofstorage locations in the memory. A reset pulse will arising from theprimary switch driver 89 of FIG. 5 causes the flip-flop device 194 torevert to a LO state when the memory functions are changed. A logical HIvalue on the output of AND device 198 corresponds to deglitch conditionsbeing achieved.

ALTERNATIVES TO THE PREFERRED EMBODIMENT

While the various elements, groups and circuits shown in the preferredembodiment of the preceeding FIGS. 3 through 5 and FIGS. 7 and 8 providea sufficient system for performing the generalized signal processingsubject to this invention, it should be readily apparent that optionalcircuitry can be used to accomplish the various functions. Further,future selection and utilization of the various alternatives is subjectto the direction and development of generic technologies. Severalexamples employing alternative contemporary technologies to the abovedescribed preferred embodiment are presented.

The memory devices used in the several delay lines 83, 108, 130 and 144are discrete-time analog devices which could be replaced by digitalstorage elements and the appropriate analog-to-digital conversionelements. The inverse conversion of the digital-to-analog signal wouldalso be required. The band splitting functions which are indicated byanalog band filters could be accomplished by digital filteringtechniques such as finite impulse response (FIR) or other transversaldigital filters. The adaptive noise cancelling filter of FIG. 7 can beaccomplished in a variety of ways from very simple phase-locked-loopcircuits to the use of digital computation on the input signal usingpreprogrammed pattern comparison algorithms. The nonlinear amplifierelement demonstrated in FIG. 5 is a double sided logarithmic amplifier,however many other amplifiers could serve appropriately. In this regardthe first stage of a recording enhancing device (commonly called"Dolby") such as a Signetics Corp. NE570N could serve adequately.Finally, a variety of circuit noise reduction schemes such as parallelprocessing of inverted signals (common mode rejection) mightadvantageously be employed. The various advantages of the aforementionedalternatives would be determined as much by cost and component countfactors as by technical improvement of the signal processing quality tobe accomplished. In this regard the aforementioned alternatives areincluded herein for the sake of completeness, and such improvements incircuitry should be obvious to those skilled in the application ofdigital and analog electronic integrated circuitry.

RELATIONSHIP TO THE PHYSIOLOGICAL PROCESS OF HEARING

Referring now to the various FIGS. and particularly to FIG. 3,attributes of the invention will be demonstrated by describing thefunction of each element as it relates to the enhancement of orsubstitution for the variety of lost physiological processescharacteristic of hearing deficiency.

In operation the auditory signal is transduced into an electrical signalby the microphone 40 and the buffer amplifier 42 in a conventionalmanner. The adaptive noise cancelling element 44 provides a means forseparating the meaningful portion of the auditory signal from noise, afunction which is normally provided by the neural feedback in the ear.The obfuscation of meaningful information by the addition of competingsignals is called masking, and in patients having any neural hearingdeficit, the ability to derive information in the presence of masking iscompromised. The placement of the adaptive noise cancelling element 44near the front end of the signal processing therefore provides anautomatic means for compensating for lost masking definition.

The next element in the illustrated signal processing series is anautomatic gain control element 45, and it serves the purpose ofvariously enhancing and suppressing the energy of the acoustic signal.Its physiological counterpart in the auditory system is bones of themiddle ear, and also in the neural threshold alteration which occurs atthe mechanical-to-neural transduction site. In some particularindividuals the automatic gain control element 45 also would findapplication to severely retard signal strength due to hyper-sensitivityto signals which occur as a result of depressed neural thresholds fromcertain pathologies of cellular electrolytes.

The band separating network 46 provides the means for directing specificfrequency ranges of the auditory signal through the processing of theremaining signal modification networks 50. Since the perception of theauditory signal occurs on a frequency-by-frequency basis, and becausethe loss of hearing is directly related to the failure of certain cellswhich are responsible for certain frequencies, any alteration of theauditory signal must also be performed in recognition of the uniquefrequency/pathology relationship. The band separating network 46performs such a function. The specific number of bands is dependent uponthe pathology of the individual and also is limited to an upper numberby linguistic considerations. For example an individual having simplehigh frequency loss might only require severe modification of thefrequencies contained in a single frequency band from 2500 to 3500 Hzwhereas another individual with the capacity to hear only ten distincttones over the entire auditory range might require ten bands, with eachgenerating the single audible tone.

The signal which is transmitted in each selected band is directedthrough a signal modifying network 50. Within each of these networks arefour distinct elements for performing specific signal alteration inorder to optimize the transmission of information contained in thesignal over those neural structures which remain viable. The bandlimitedsignal is caused to be processed through a nonlinear amplifier element.This element provides for the greater amplification of small amplitudesof the signal while suppressing the larger amplitudes. The effect ofsuch processing tends to promote the effect of sound energies of lowpower which have significant information content such as the consonants"v" and "f". Such singular devices have been variously called speechcompressors or peak clippers. The detection of the crossing of zero isof much greater importance than is the determination of the amplitude ofthe vibrating source as far as the determination of linguisticallysignificant events is concerned.

After the band limited signal has been caused to pass through thenonlinear amplifier element 52, it enters the harmonic frequencytransposer element 54. In the harmonic frequency transposer element theband limited signal is shifted in frequency, while retaining all of theharmonic relationships in the original signal. Applicant believes thatsuch harmonic retention is significant to the optimization of neuralcues for recognizing auditory patterns. The effect of the shiftingprocess is to redirect portions of the auditory signal from neurallydead frequencies to those neural channels which remain viable to thetransmission of particular frequencies. Shifts which are made in thismanner should be performed only enough to cause enhanced neural cuessince distortion of the auditory signal rises significantly with largeshifts. However, there is far greater tolerance of shifts in thefrequencies above about 2000 Hz than there is in the lower frequencies.

The bandfilter element 55 which receives the transposed bandlimitedsignal serves to limit the energy which will drop below the lowerfrequency limit of the original band limited signal. The bandfilterelement 55 may also be used in a narrow band manner to provide anacoustic cue representing the entire energy content of the band forthose cases where an individual retains only sensitivity for a few tonesin the entire auditory spectrum. The signal which is directed intobandfilter elements 55 emerges from through a gainshaping amplifier 56to provide a final degree of tuning to the signal. This last mentionedelement 55 may be used to compensate for subtle differences in thesensitivities of neural hair cells and structures to mechanicalexcitation.

While the elements represented are required for the most general case ofsignal processing, their function may be so adjusted to provide no neteffect if it is not required. Similarly, the sequence of the functionsmay also be performed in other than the order described herein.

The separately processed signal bands 50 are next summed together withvarious intensity weightings being applied by potentiometers 58 atsumming amplifier 59. It is necessary that equivalent time delays beprovided in each of the signal bands prior to their recombination.

It is readily apparent that if all elements in any of the signalmodifying networks 50 except the frequency transposer element 54 areshunted with regard to the signal path, and if both clock `S` 86 andclock `F` 87 rates are made equal, a pure delay is imposed on thatportion of signal traversing said network. By further requiring that allof the clock `S` 86 rates be made equal in all of the individual signalmodifiying networks 50, the said delayed signals would provide for thetemporal or phase matching of all signals processed in the plurality ofsignal modifying networks 50 of FIG. 3.

So similarly may other combined sets of parameters in the variouselements provide yet other special cases. For example, the total energycontent of an entire frequency band may be represented as a narrowbandsignal of harmonically related information by a suitable choice oftransposition, filtering, and heterodyning.

DIAGNOSIS AND TREATMENT OF HEARING LOSS

Diagnostic methods for the treatment of hearing loss employing the meansrecited above require a significant departure from contemporaryaudiological practice.

In contemporary audiology, the pure tone audiogram is a primary methodfor evaluating auditory function. This evaluation is performed bypresenting the listener with a single-frequency signal and incrementallyincreasing or decreasing the intensity of the signal. The listenerindicates when the tone is audible, and when he indicates that the soundis not audible, a threshold of hearing is established for that tone.Then the process is repeated at typically six or eight more frequenciesspaced at approximately one octave intervals. At the conclusion of thetest, an "audiogram" is constructed by joining the test points. It iscommon practice for these straight line segments to be drawn on eithersemilogarithmic or double logarithmic coordinates. These veryqualitative graphs are generally interpreted as a representation of thefilter characteristics of the listener's auditory system.

A substantially more accurate characterization of the auditory systemfilter characterization is obtained by presenting the listener with aconstant intensity signal which is caused to vary monotonically infrequency. As the signal slowly sweeps across the sonic frequency range,it is progressively presented to individual hair cell groups. As long asthere is a normal neural structure and function, the listener respondsto appropriate threshold intensity levels. When pathologies affect aparticular site along the basilar membrane, a sudden "dropout" of thethreshold sensitivity occurs, and the listener fails to respond. Theintensity level is subsequently raised, and a second identification ofthe "dropout" is obtained. The procedure is repeated at higher intensitylevels, with the frequency ranges of the sweeps narrowed to minimizeuncomfortably loud signals from being addressed to normal portions ofthe mechano-neural network.

The recording of the listener's response can be automated using aconventional X-Y plotter which has a pen lifting device. The intensitylevel is represented as a voltage on the "Y" input to the plotter, andthe frequency is represented as a voltage on the "X" axis to theplotter. The patient causes the pen of the plotter to lift by depressingan electrical switch. As the sound is administered to the patient in aslowly increasing frequency, a constant intensity line will be drawnacross the plotter. In those regions where hearing is lost, the pen willbe lifted, and the line will be caused to discontinue. Then theintensity is increased, and a second line is drawn. This process iscontinued until a filter characteristic of the patient's auditory systemis established.

The sweeping of frequency at a constant intensity is a most importantalteration to the generation of a filter characterization of the ear.First, since hearing is sensed on a frequency by frequency basis, themethod permits frequency resolution of the "filter" not otherwisepossible. Second, the method minimizes "recruitment" effects on thefilter characterization. Applicant postulates that recruitment, acondition in which a sudden onset of hearing occurs upon thepresentation of a gradually increasing stimulus intensity, arises whenhair cells adjacent to regions of pathology are suddenly brought abovetheir transmitting threshold. Finally, the method enables theidentification of the very narrow bands of frequency reception which maybe present in persons who are profoundly deaf to pure tone stimuli whichare not addressed to the viable frequencies.

The accurate characterization of the listener's auditory channelspermits a ranging of the variable parameters of a master generalizedsignal processing hearing aid. The number of separate channels which areused for the signal processing is dependent upon the number of dropoutsin the the listener's frequency response, their position along thefrequency axis, and their magnitudes. The number and frequency positionof the separated channels is also influenced by the phonemicrequirements of language.

After the initial signal processing channels are determined using theswept frequency audiogram, an initial set of values are chosen for thevarious gains and gainshaping elements. The listener is then presentedwith well known word discrimination lists, i.e. spoken words which areknown to differ in subtle ways from other words, and his test scores aretaken. On the basis of the types of words that are missed and thespectral content of those words, the appropriate gain changes, and ifnecessary transposition placements and characters are altered. Thelistener is also an active participant in this alteration of parametersportion of hearing aid prescription.

Next, a competing noise signal is added to the background against whichthe listener is presented the word discrimination tests. Thecharacteristics of the noise cancelling and automatic gain controlportions of the master hearing aid are adjusted to provide a maximalresponse to these tests. Again the parameter adjustment is made with theactive participation of the listener in the adjusting procedures.

A fine tuning process is the final step in determining the generalsignal processing required for an individual listener. During this finalphase, a series of pre-recorded words representing a variety of adultmale, adult female and child's voices are presented. The ranging ofparameters responsive to distinguishing language spoken by these voicesis determined, and these parameters become the final prescription for anindividual hearing aid.

While a specific embodiment and method for the purpose of the inventionhas been disclosed, and alternatives to certain elements and procedureshave been suggested, it will be appreciated that although each of theillustrated elements and suggested steps is desirable for the mostgeneralized signal processing, some of these elements or steps may be soadjusted to produce no net effect, or in special cases, some elements orsteps may be eliminated entirely. Similarly, it will be appreciated thatthe sequence of individual signal processing elements also may bepermuted to address individual cases. It will be further appreciatedthat other circuits to implement the elemental functions will be knownto those skilled in the art and are to be understood as within the scopeof the invention as defined in the appended claims.

I claim:
 1. A method of providing a quantitative measure of a hearingdefect of a patient in relationship to generalized signal processingparameters, comprising the steps of:(a) determining and recordingfrequency bands to which the patient's auditory system responds at apredetermined level and frequency bands to which the patient's auditorysystem is unable to respond at the predetermined level through theagency of a swept frequency tone generator operated at incrementalintensity levels; (b) providing a number of frequency band limitedsignal processing paths in accordance with said determining andrecording step, the processing paths being operative to generate asignal responsive to those frequency bands to which the patient'sauditory system responds at the predetermined level; (c) adjusting andrecording the parameters necessary to shift in a harmonic sustainingmanner the frequency bands in which the patient's auditory system doesnot respond to frequency bands in which the patient's auditory systemdoes respond; and (d) adjusting and recording within each frequency bandprocessing path and in the aggregrate of all said frequency bandprocessing paths at least such parameters as are necessary to shape,linearly or nonlinearly amplify, amplitude compress, and adaptivelyfilter background noise from the signals generated by the processingpaths in a continuously variable and adaptively fashion so as to producean aurally recognizable signal to the patient.
 2. A method as recited inclaim 1 wherein the step of recording the patient's responsive andnon-responsive frequency bands includes the step of requiring thepatient to displace an external transducer in response to saidnon-responsive frequency band.
 3. A method as recited in claim 2 whereinthe recording of the patient's responsive and non-responsive frequencybands includes the step of operating a plotting device which isresponsive to the generated signal's intensity and to the displacementof the external transducer.